一种稳健的自适应波束形成器

当信号噪声比超过一定的门限时,线性约束自适应波束形成器对天线的幅相误差有很高的敏感度,即使在误差很小的情况下,期望信号也会如同干扰一样被抑制掉。该文通过对广义旁瓣相消器的阻塞矩阵加以改进,提出了一种对阵列天线误差有良好稳健性的自适应波束形成器。该方法基于广义旁瓣相消器结构,可方便地进行部分自适应,降低运算量。     

一种改进的基于LC-GSC的主瓣干扰抑制方法

针对线性约束广义旁瓣相消器(LC-GSC)存在主瓣干扰时天线自适应方向图会出现主瓣波束畸变及副瓣电平升高的问题,给出了一种通过阻塞矩阵对数据进行预处理的主瓣干扰抑制方法,对线性约束广义旁瓣相消器进行了改进,先利用阻塞矩阵预处理实现主瓣干扰抑制,再通过线性约束广义旁瓣相消器进行波束形成,有效地解决了波束主瓣变形及旁瓣电平升高的问题。仿真分析验证了改进方法的有效性。     

基于二阶锥规划的相干信号深零陷自适应波束形成

针对常规自适应波束形成算法在强相干干扰情况下零陷深度不够、甚至干扰抑制失效的问题,提出了一种基于二阶锥规划的相干信号深零陷自适应波束形成算法。该算法首先对接收数据协方差矩阵进行Toeplitz重构,使其包含信号和干扰的所有方位信息。然后重构了干扰加噪声协方差矩阵。接着在保证期望方向波束无失真前提下,约束主瓣宽度和旁瓣电平,使得波束形成器干扰和噪声的输出功率最小。最后将该问题转化为凸优化中的二阶锥规划问题,并使用凸优化工具箱进行快速求解。仿真结果验证了该算法的有效性。     

基于子带TF-GSC麦克风阵列语音增强

为了加快基于传递函数广义旁瓣相消器的麦克风阵列语音增强系统的收敛速度,将其自适应模块的输入信号分解到子带以进行处理,并将多通道维纳滤波器引入传递函数广义旁瓣相消器的非自适应支路,以便更有效地抑制非相干噪声.实际测试结果表明,相对于基于全带广义旁瓣相消器的麦克风阵列语音增强系统,采用该子带传递函数广义旁瓣相消器结构的语音增强系统具有更高的输出信噪比.     

一种基于广义旁瓣相消的稳健降维方法

在自适应阵列信号处理中,为了高效进行干扰和噪声对消,提出了一种基于广义旁瓣相消的稳健降维方法。该方法利用信号子空间特征矢量和期望信号的投影导向矢量构造稳健降维阻塞矩阵,阻塞期望信号和噪声分量,使辅助支路中只含有干扰信号,达到了降维的效果,同时提高了算法对阵列天线误差的稳健性。仿真结果表明,该方法对阵列模型误差不敏感,可以有效地降低运算时间。     

时域宽带自适应波束形成器设计

针对现有含反馈支路的时域宽带自适应波束形成器运算量大、干扰抑制能力受限的问题,基于广义旁瓣相消结构,引入一条固定系数的全极点反馈支路,设计一种新的时域宽带自适应波束形成器,减少了前向支路所需抽头延迟线个数即自适应权个数,从而降低了运算量,加快了收敛速度。全极点反馈支路以逼近包含干扰频带的带通滤波器为目标进行离线最优化设计,在保证稳定性的同时,增强了波束形成器的干扰抑制能力。仿真结果表明：与现存含反馈支路的时域宽带自适应波束形成器相比,采用相同的自适应算法,设计的波束形成器收敛更快、干扰抑制能力更强;实现相同的 SINR 改善时,新的波束形成器所需运算量远小于现存波束形成器。     

一种稳健的自适应波束形成算法

一般自适应波束形成的主瓣和零陷比较尖锐,当存在指向误差,干扰信号或天线平台快速运动时,若实时处理速度跟不上,将导致干扰抑制能力下降,采用高速处理算法代价太大.本文研究了利用导数约束法来改进广义旁瓣相消器的稳健性,对信号的自相关矩阵进行改进,使主瓣和零陷展宽,仿真试验表明该方法对指向误差和当干扰或天线平台快速运动时都具有很好的稳健性.     

基于广义旁瓣相消的新降维方法

针对阵列信号中部分自适应信号处理问题，提出了一种新的基于广义旁瓣相消的降维方法。该方法利用波束形成无法对加性白噪声进行抑制的特点，根据广义旁瓣相消器的结构，设计出一种新的阻塞降维矩阵，使辅助通道内仅存干扰信号，从而增强主辅通道关于干扰信号的相关性，更有利于抑制干扰，同时也达到了降维的效果。另一方面，在存在阵列误差的情况下，借助稳健的波束形成算法，对阻塞降维矩阵进行校正，既使算法对模型误差不敏感，又便于进行降维后的部分自适应处理，降低运算量。最后通过大量的计算机仿真验证了所提方法的有效性和优越性。     

基于GSC结构的波束形成器的互耦校正

广义旁瓣对消器(GSC)是线性约束最小方差波束形成器的等效结构。但由于实际互耦误差的影响,GSC的支路会出现期望信号泄漏,导致此波束形成器不能正常工作。文中基于一种新定义的互感矩阵,对阵元之间的互耦进行精确的量化和分析,提出了一种GSC波束形成器的互耦补偿方法。该方法通过修改GSC的参数,解决了期望信号相消的问题。仿真试验表明,该方法能精确的校正阵列天线阵元间的互耦效应,使GSC波束形成器对互耦误差具有稳健性。     

基于最坏情况性能优化的稳健自适应宽带波束形成

针对期望信号导向矢量失配会导致自适应波束形成算法的性能急剧下降这一问题,提出了一种新的基于最坏情况性能优化的稳健自适应宽带波束形成方法。该方法通过将实际期望信号的导向矢量约束于不确定集合中,来提高算法的稳健性;同时增加阵列空间响应偏差的二次型约束,可避免引起信号有较大失真。计算机仿真结果表明:该方法不仅能够抑制导向矢量偏差所带来的影响,有效控制波束主瓣区域内信号的畸变和旁瓣区域电平,还改善了阵列输出的信干噪比,使其更接近最优值。     

面向二元麦克风小阵列改进的广义旁瓣抵消器语音增强算法

二元麦克风小阵列在手机、助听器等受空间、成本以及运算能力限制的设备中被广泛研究用以提高目标语音质量。二元麦克风小阵列中语音增强算法主要包括波束形成方法以及相干性滤波器方法。波束形成方法的思想是利用目标声源相对阵列的位置关系获取相应的时域和空域信息,可以保留目标声源方向的信号而抑制其他方向的干扰信号;相干性滤波器方法则通过阵元间不同信号的相关性进行噪音抑制。考虑这两种类型方法的优点,本文提出一种面向二元麦克风小阵列改进的广义旁瓣抵消器语音增强算法,通过在广义旁瓣抵消器的固定波束形成支路上使用相干性滤波器,提高固定波束形成输出信号的信噪比,然后在广义旁瓣抵消器自适应支路利用阵列的时域和空域信息对固定波束形成支路输出的信号中残余噪音进行估计,进而获得增强后目标输出信号。仿真和实际试验表明,本文提出的算法明显优于单独使用小阵列波束形成算法和相干性滤波器算法。      

Effects of desired signal on the performance of a sidelobecanceller

The performance of a sidelobe canceller is studied in a communication system where the desired signal is continually present and its amplitude in the auxiliary antennas may be large (above thermal noise). In such a system to avoid the cancellation of the desired signal, one can use a reference signal and adjust the auxiliary antenna weights to minimize the error between the reference signal and the sidelobe canceller output or one can use a steering vector. It is shown that in spite of the reference signal or a steering vector, the presence of the desired signal in the auxiliary antennas degrades the interference suppression provided by a sidelobe canceller. The amount of degradation depends on the number of degrees of freedom, the strength of the various signals, and the angular resolution of the auxiliary antenna array. Signal scenarios involving a single interfering signal as well as those involving multiple interfering signals are considered     

空间变迹法在合成孔径声呐成像处理中的应用

空间变迹法（SVA）作为一种超分辨率图像处理技术,能在抑制旁瓣的同时保持良好的主瓣分辨率。将空间变迹法引入合成孔径声呐（SAS）成像,比传统的加窗处理能获得更好的成像效果。对标准空间变迹法进行了推广,得到非标准奈奎斯特采样率下的通用算法,并应用于SAS成像处理。仿真结果表明,二维通用算法在任意奈奎斯特采样倍数时．均能在不增加主瓣宽度的同时得到良好的旁瓣抑制效果,提升了SAS成像分辨率。     

A Robust Space-Time Generalized Sidelobe Canceller

This paper deals with adaptive array beamforming based on a two-dimensional generalized sidelobe canceller (GSC) with robust capability. It has been known that the performance of conventional GSC is quite sensitive even to pointing error. In conjunction with the joint desired user’s code-aid and signal subspace estimating approach, the proper quiescent weight vector and blocking matrix of GSC can be obtained to suppress the leakage of desired signal. However, space signature is estimated by exploiting the knowledge of the spreading code of the user of interest and the orthogonality between noise and signal subspaces. In this paper, the space-time signature estimation can be integrated jointly. Therefore, the desired signal cancellation does not occur; even the pointing error is relatively large. Computer simulation results show the effectiveness of the proposed approach.     

Adaptive beamforming with the generalized sidelobe canceller in the presence of array imperfections

Antenna designers often employ linearly constrained adaptive beamforming as an antijamming measure. With minimal a priori knowledge of the signal environment, this technique nulls out jammers while simultaneously preserving the quality of the main lobe so that a friendly look-direction signal can be received with unity gain. Unfortunately, in the absence of special strategies, linearly constrained adaptive beamforming is hypersensitive to array imperfections when the input signal-to-noise ratio exceeds a certain threshold. This hypersensitivity manifests itself as a nailing of the friendly signal as if it were a jammer. Luckily, the signal nulling problem can be easily remedied by artificial receiver noise injection. A particularly simple and general structure for linearly constrained adaptive beamforming was proposed during the 1970's, and is known as the generalized sidelobe canceller. A detailed analysis of the generalized sidelobe canceller in the presence of array imperfections is discussed, and two new artificial receiver noise injection algorithms are proposed. Computer simulations are included to demonstrate that use of these new algorithms alleviates the signal nailing problem without seriously compromising jammer nulling. For the special case of the Capon maximum-likelihood beamformer, simple approximations are presented for: 1) the Wiener output signal-to-interference-plus-noise ratio (SINR_{0}astr), 2) the antenna element error variance that causes a 3 dB loss ofSINR_{0}astrfrom its value for an ideal array, and 3) the optimal artificial receiver noise that maximizesSINR_{0}astr.     

基于预失真编码激励与最小方差波束形成的超声成像算法

为了提高医学超声成像的轴向分辨力,文中提出一种基于预失真编码激励与改进最小方差自适应波束相结合的成像方法。该方法一方面对编码激励信号进行预失真和旁瓣抑制处理,补偿探头对信号的影响,提高发射端的分辨率与成像对比度;另一方面对接收数据采用改进的最小方差波束形成,有效提高了医学超声成像过程中的分辨率并改善自适应算法的鲁棒性。仿真结果表明,相对于恒包络信号与传统的延时叠加波束形成算法,文中算法从信号源和接收数据两方面提高了成像的对比度和分辨力,为实现高质量的超声成像系统提供了理论依据。     

基于稀疏约束和SRV约束的宽带自适应波束形成

针对传统的宽带MVDR自适应波束形成中,抑制干扰的同时会抬高旁瓣电平,且过多的线性约束会导致波束输出的SINR性能下降的问题,提出了一种基于SRV约束和稀疏约束的低旁瓣、高增益宽带自适应波束形成方法.该方法在窄带MVDR自适应波束形成基础上,通过增加波束图稀疏约束来降低波束的旁瓣电平,同时利用空间响应偏差(SRV)约束将窄带算法推广到宽带MVDR 自适应波束形成中,极大地降低了算法的复杂度,改善了波束输出的SINR 性能.与传统方法相比,该方法在降低宽带波束的旁瓣电平的同时,还具有良好的干扰抑制效果.数值仿真实验验证了该方法的有效性.     

Design of partially adaptive array beamformers based on informationtheoretic criteria

In this paper, two basic problems in designing partially adaptive array beamformers based on the structure of generalized sidelobe canceller (GSC) are considered. The first problem is to decide the proper dimension of the required adaptive weight vector. Using the information of the array output power, we develop the detection formulas for the information theoretic criteria AIC and MDL to decide the proper dimension of the adaptive weight vector. If the input noise power is unknown a priori, efficient methods are proposed for estimating the input noise power in both cases, with and without the desired signal, to make the detection formulas still feasible. The second problem is to find the most appropriate channel signals for weight adaptation for efficiently canceling interference. An efficient method based on the maximum power reduction criterion is presented for selecting the most desired channel signals from the output of the signal blocking matrix. Theoretical analysis concerning the performance of the proposed methods is made. Computer simulations showing the effectiveness of the proposed methods are also provided     

A robust source tracking algorithm for partially adaptive arrays

A new algorithm for overcoming the problem of incidental cancellation and improving the convergence behaviour of large partially adaptive arrays is proposed and investigated. The algorithm uses an auxiliary adaptive source tracker to separate individual sources in the environment before using the resulting outputs to cancel unwanted jamming picked up through the main beam of the system. Because the source tracker's outputs are roughly orthogonal in the steady state and the positions of sources of significant powers are available, the algorithm possesses fast convergence behaviour and solves the problem of incidental cancellation in the most natural manner by adding back the desired signal to the canceller output if the former is cancelled during the adaptation process. In addition, the algorithm has about the same computational complexity as partially adaptive generalized sidelobe cancellers using steepest descent based algorithms.     

Two‐Microphone Generalized Sidelobe Canceller with Post‐Filter Based Speech Enhancement in Composite Noise

This paper describes an algorithm to suppress composite noise in a two‐microphone speech enhancement system for robust hands‐free speech communication. The proposed algorithm has four stages. The first stage estimates the power spectral density of the residual stationary noise, which is based on the detection of nonstationary signal‐dominant time‐frequency bins (TFBs) at the generalized sidelobe canceller output. Second, speech‐dominant TFBs are identified among the previously detected nonstationary signal‐dominant TFBs, and power spectral densities of speech and residual nonstationary noise are estimated. In the final stage, the bin‐wise output signal‐to‐noise ratio is obtained with these power estimates and a Wiener post‐filter is constructed to attenuate the residual noise. Compared to the conventional beamforming and post‐filter algorithms, the proposed speech enhancement algorithm shows significant performance improvement in terms of perceptual evaluation of speech quality.